CleanCutAudio

Audio Trimming and DSP Tool — the complete user manual: audio editing, the DSP chain, loudness metering and exporting, explained step by step.
📖 How-to guide + technical deep dives 🎧 Audio editor for Windows 🏷️ Version 2.2.0.0

①Introduction

What CleanCutAudio is and who this manual is for.

CleanCutAudio is an audio editor for Windows designed to clean up, polish and “master” recordings: voice, podcasts, music, samples. It works with MP3 and WAV files and offers three groups of tools: a waveform editor (cuts, fades, silence, reverse, pitch), a professional DSP effects chain (noise reduction, de-esser, equalizer, compressor, ambient, soft clipper, gain, stereo widener) and a set of analysis tools (LUFS/loudness, FFT spectrum analyzer, VU meters, mastering report).

This manual is meant to be read on two levels. The main text explains what each feature does and how to use it, in plain language. The blue-gray boxes marked ⚙️ Technical deep dive add details on the parameters (thresholds, ratio, LUFS, true peak…) for those who want to understand what happens “under the hood”. If you are a beginner you can safely skip the technical boxes: the rest of the text is all you need to get to work.

✂️

Non-destructive editing

Cuts, regions, fades and silence applied on the waveform with draggable markers.

🎛️

DSP chain per region

Eight modules in series, with ready-made presets and real-time preview while you listen.

📊

Professional metering

LUFS loudness to EBU R128, true peak, FFT spectrum and mastering report.

💾

Flexible export

MP3 (CBR/VBR) and WAV, per region or whole file, with ID3 tags.

Conventions In this manual keys are written like this: Space, Ctrl+R. Menu and button names are in bold. “Region” means a portion of audio bounded by cut markers; “selection” means the range between the start marker and the end marker.

②The interface

How the main window is organized (v2.2.0.0).

CleanCutAudio main window with a file loaded
The main window with a track loaded: on the left the file browser and file data, in the center the stereo waveform (left channel in blue, right in red) with selection and fades, on the right the quick buttons and the two radial VU meters.

At startup, until you load a file, the waveform shows a welcome screen inviting you to drag in an MP3 or WAV, plus a reminder of the main shortcuts. The window is divided into a few fixed areas:

AreaWhat it does
Menu barAt the top: File · Project · Edit · Cuts&Regions · Processing · Tools · Voice&Text · Audio Device · Settings · Help (see the table below).
File browserTree panel with your computer's folders, favorites and quick locations (C:\, Music, Desktop, Documents), a Change Directory button and a Text Search (F3 = Next) box. Double-click a file or press Enter to load it.
Information panelTop left: MP3 Loaded Data (file name, folder, duration, sample rate, channels, bit depth, total samples, bitrate, size) and ID3 Data (title, artist, year).
WaveformThe central area where you view and edit the audio: left channel (blue) and right channel (red), markers, regions, fades, playhead (cyan line), time ruler. The portions outside the selection are dimmed and marked REMOVED SECTION.
Quick buttons (right)DSP, Normalize LUFS, Sound Analyzer, STT Transcribe, STT Transcript, TTS TextToSpeech: shortcuts to the main windows and tools.
VU metersTwo radial Left / Right meters on the right, showing the level in real time.
Transport (bottom)Play (green), Stop (red), Record (recording), PlayHead readout with time and percentage, Enable Loop and Follow PlayHead checkboxes. At the very bottom, the status bar shows the file path and the active audio driver (e.g. WASAPI (…) – Shared).

The menu bar

MenuMain items
File📂 Open… · 💾 Save As… · ✏️ Edit mp3 Info…
Project📦 Open Project… · 📥 Save Project… · 💾 Save Project As Single File…
EditInsert Silence · Reverse · Cut/Copy/Paste · Paste File At Playhead · Pitch Shift… · Fix DC Offset · Trim Start/End Silence · Trim Internal Silence · Auto De-Click · Trim to Selection · Trim to Active Region · Undo
Cuts&RegionsInsert Marker Cut · Delete Marker Cut · Remove All Markers Cut · Trim to Active Region
ProcessingRealTimeDSP… (the DSP chain) · Sound Analyzer…
ToolsLoudness Normalization (LUFS)… · Recorder… · Tone Generator…
Voice&TextText-to-Speech Module ▸ System Voice… · Speech-to-Text Module ▸ Convert Speech to Text / View Transcript Form…
Audio DeviceChoice of audio driver and device.
SettingsExport Settings… · DSP Settings…
HelpAbout…

Many features also have a quick button in the right-hand column or a keyboard shortcut. The DSP window (effects chain) and the Sound Analyzer open separately and can stay open while you work.

Technical deep dive The interface uses a high-readability dark theme and is DPI-aware: it scales correctly with Windows set to 125%, 150% or 200%. Window position, panel proportions and DSP module state are saved on exit and restored on the next launch.

③Opening a file

Three ways to bring audio into the editor.

  1. Drag and drop — take an MP3 or WAV file from the desktop or File Explorer and drop it onto the waveform.
  2. From the built-in browser — move through the folders in the tree panel on the left, then double-click the file (or select it and press Enter, or right-click ▸ Load into Waveform).
  3. From the menu — File ▸ 📂 Open… and pick the file in the dialog.

In the browser you can add the folders you use often to your Favorites (right-click ▸ Add to Favorites) and jump quickly to standard locations with the roots button. Once loaded, the file appears in the waveform and its data shows up in the information panel.

Tip Before heavy editing, consider working on a copy: many editing operations (cut, silence, reverse, pitch) change the audio in memory. You can always undo with Edit ▸ Undo, and the original on disk is not touched until you use Save As or export.

⑤Selection, markers and regions

The heart of editing: marking out the portions you want to work on.

CleanCutAudio distinguishes two concepts that are best kept separate:

  • Selection — the range between the start marker and the end marker. It is the portion used by loop playback, transcription, LUFS analysis of the area and commands such as “Trim to Selection”.
  • Regions — portions of audio created by inserting cut markers. Each region has a name, a color, its own fades and — above all — its own independent DSP chain.

Start and end markers

Drag the two markers directly on the waveform to define the selection. During playback the selection stays consistent: if you move the markers, the playback position is brought back inside the range.

Cut markers and regions

  1. Place the position where you want to split and press + to insert a cut marker (or Cuts/Regions ▸ Insert Marker Cut).
  2. Each cut marker splits the audio into consecutive regions. Add as many as you need to isolate the parts (the tracks on one side of a cassette, the songs in a mix, the speakers in an interview…).
  3. Remove a single cut with − (or Delete Marker Cut); remove them all with Ctrl+Shift+X (or Remove All Markers Cut).

Click a region to select it: it becomes the “active” region, the one the DSP window and several commands refer to. Press F10 to show/hide the list of regions and cuts.

Technical deep dive Each region carries its own complete, independent DSP configuration (DspConfig). This means you can apply, for example, heavy noise reduction only to the spoken intro and different compression to the chorus, all in the same file. On export, each region is rendered with its own chain.
Tip Cut markers take “priority” when you click near them: if you have trouble selecting a region because the click keeps grabbing the cut, move a few pixels away from the marker before clicking.

✂️Marker Cut: the complete guide

The central concept of CleanCutAudio — where regions come from and everything that follows from them.

Waveform split into regions with the TIMELINE overlay
The same track split into four regions. The red brackets are the start/end markers (MARKER START/END), the orange lines are the cut markers (CUT). At the bottom, the Region 01…04 labels; the selected region is highlighted. On the right, the TIMELINE overlay (key F10) with the exact times of every marker, region and cut.

The cut marker (marker cut) is the element the program's entire workflow revolves around. It is not just a decorative “notch”: each marker cut divides the audio into regions and each region becomes a self-contained block, with its own fades, its own DSP chain and its own export behavior. Understanding marker cuts well means understanding how CleanCutAudio thinks.

What a marker cut is and what it creates

When you insert a marker cut, the audio to the left and to the right of the cut becomes two separate regions. Adding more gives you a sequence of consecutive regions covering the whole file. Each region carries:

🎨

Name and color

Each region has a name and a color, so you can recognize it at a glance on the waveform and in the lists.

🎚️

Its own DSP chain

Noise reduction, EQ, compressor… all modules have independent settings for each region.

🔉

Its own fades

Fade-in and fade-out are defined per region, with dedicated handles.

💾

Independent export

On export, each region is rendered with its own chain and its own range.

Inserting, moving and removing marker cuts

OperationHow to do itResult
Insert a cut+ at the desired point, or Cuts/Regions ▸ Insert Marker CutThe audio is split into two regions at the cut point.
Move a cutDrag the marker cut on the waveformThe adjacent regions resize accordingly, in real time.
Remove a single cut−, or Cuts/Regions ▸ Delete Marker CutThe two regions on either side merge back into one.
Remove all cutsCtrl+Shift+X, or Remove All Markers CutYou are back to a single region (the main region).
Show/hide the TIMELINE overlayF10Shows the chronological list of markers, regions and cuts with their exact times.
“Smart” selection When you click near a marker cut, the program automatically selects the region the cut belongs to as well: this is the natural behavior for working on that portion right away. If instead you want to select the region in the middle without grabbing the cut, click a few pixels away from the marker.

The TIMELINE overlay (F10)

Pressing F10 brings up the TIMELINE overlay, the chronological map of the file: it lists, in order, MARKER START, each REGION with its range (e.g. Region 03 [1:46.237 → 2:27.690]), each CUT with its position and finally MARKER END. Regions are named automatically (Region 01, 02, …) and cuts stay “(unnamed)” until you rename them; the entry for the selected region is highlighted. It is the most accurate way to check cut points down to the millisecond.

The active region

Clicking a region makes it active: it is the region the DSP window (which shows the region's name and color at the top) and several editing commands refer to. In other words, to apply effects or actions to a specific portion, first you select it, then you act. Hovering the mouse over regions highlights them without activating them.

What follows from marker cuts

1 · DSP processing for different parts

This is the most powerful consequence: since each region can have an independent DSP chain, you can treat different parts of the same file differently. Typical examples: heavy noise cleanup only on the spoken intro; more compression on the chorus; brighter highs only on a poorly recorded verse; turning the de-esser off where it is not needed.

2 · Clean fades and transitions

Each region has its own fade-in/fade-out handles. When you drag the marker cut between two regions, the program prevents the fades of neighboring regions from overlapping beyond the allowed limit, so the join point stays free of clicks and gaps. It is the correct way to join two tracks or split a medley.

3 · Trim to region

With Cuts/Regions ▸ Trim to Active Region (or Edit ▸ Trim to Active Region) you discard everything else and keep only the selected region: useful for extracting a single song from a long recording split with markers.

4 · Multi-region export

On export, the summary lists each region with its range and its DSP chain. With no cuts, the single main region is exported (the range between the start/end markers); with several cuts, each region is rendered with its own settings.

Technical deep dive Each region (AudioRegion) contains its own boundary samples (StartSample/EndSample), fade durations in seconds, color, name and a complete DSP configuration (DspConfig). Moving a marker cut updates the boundaries of the adjacent regions in real time, with a check that blocks the move if the fades would overlap (fadeIn + margin ≥ fadeOut). Marker cuts have hit-test priority over regions: when the pointer is within a few pixels of a cut, the click grabs the cut and selects its associated region.

Typical workflow with marker cuts

  1. Load the long recording (e.g. one side of a cassette with several songs, or an interview).
  2. Listen and press + at the start of each new part: you get one region per song/segment.
  3. Fine-tune the cut points by dragging the markers; adjust the fades at the transitions.
  4. Select one region at a time and set up its DSP chain (noise, EQ, compression…) in the F9 window.
  5. Export: you get each region as a separate file, each with its own processing, or you trim to a single region with Trim to Active Region.
Warning Remove All Markers Cut (Ctrl+Shift+X) deletes all cuts at once and returns the file to a single region: the splits you made are lost (you can still undo right away with Undo). Use it deliberately, when you want to start over from scratch.

⑥Fades

Smooth fade-ins and fade-outs, with no clicks.

Each region (and the main region when there are no cuts) has two fade handles: one at the top left for the fade-in and one at the top right for the fade-out. Drag them toward the center to lengthen the fade, toward the edges to shorten it. The fade-in and fade-out durations are stored for each region.

Fades are used to avoid “clicks” at the ends of a cut, to blend two adjacent regions and to end a track gracefully. When you drag a cut marker between two regions, the program prevents the fades of neighboring regions from overlapping beyond the allowed limit, so the transition stays clean.

⑦Playback

Listening to your work, with or without effects.

  • Play / Stop — the Space bar or the transport buttons. Playback starts from the playhead; if the playhead is outside the selection, it is moved back to the start of the selection.
  • Loop — enable the Loop checkbox to repeat the selection continuously: useful for adjusting an effect “by ear”.
  • Follow playhead — with this checkbox enabled, the view scrolls by itself to follow the playback position.
  • Repositioning — drag the playhead on the waveform to move the listening point, even during playback.
Technical deep dive During playback, the DSP chain of the current region is applied in real time (“live” preview). When you change a parameter or switch presets in the DSP window, the chain is rebuilt on the fly and you hear the effect immediately. This lets you fine-tune the parameters while you listen, without having to apply and listen again every time.

⑧Edit menu: editing commands

All the operations that change the audio, from the Edit menu.

These commands act on the selection or on the active region, depending on the case. You can always go back with Undo.

CommandShortcutWhat it does
Insert SilenceCtrl+WZeroes out (silences) the selected portion while keeping its duration. Useful for removing a noise or a word without shifting the rest.
ReverseCtrl+RReverses the portion (makes it play backwards).
Cut / Copy / Paste— / — / Ctrl+PCuts, copies and pastes portions of audio via the internal clipboard.
Paste File At Playhead—Inserts the contents of another audio file at the playhead position.
Fix DC Offset—Removes the DC component that shifts the waveform above or below zero (a cause of clicks and lost headroom).
Trim Start/End Silence—Removes the silence at the beginning and end of the file.
Trim Internal Silence—Shortens or removes overly long pauses within the audio (pauses between sentences, gaps).
Auto De-Click—Automatically attenuates clicks and impulsive noise (typical of vinyl or splices).
Trim to Selection—Discards everything outside the selection, keeping only the range between the two markers.
Trim to Active Region—Keeps only the currently selected region.
Undo—Undoes the last operation.
Typical voice cleanup workflow Fix DC Offset → Trim Start/End Silence to trim the ends → optionally Trim Internal Silence to tighten the pauses → Auto De-Click if there are pops → then move on to the DSP chain for noise, de-esser and equalization.

⑨Pitch Shift

Raising or lowering the pitch (Edit ▸ Pitch Shift…).

Pitch Shift changes the pitch of the current region/selection. The window offers a slider in semitones, a Preview button (listen to a preview of about 2 seconds before applying) and the Apply and Cancel buttons.

  1. Select the portion to transpose and open Edit ▸ Pitch Shift….
  2. Set the shift with the semitone slider (negative values = lower, positive = higher).
  3. Press Preview for a quick listen, then Apply to confirm.
Technical deep dive The shift is calculated in cents (hundredths of a semitone) and applied only to the portion between the markers; the rest of the file is stitched back together before and after the transposed region, updating the total length and the markers accordingly. Like all pitch-shifting algorithms, large shifts (more than a few semitones) introduce artifacts: for voice it is best to stay within a few semitones.

⑩The DSP chain

The processing modules in series, with real-time preview.

Open the window with the DSP button, from the Processing ▸ RealTimeDSP… menu or with the F9 key. The DSP chain is the set of effects the audio passes through in sequence: the output of one module is the input of the next. Each region has its own chain, so the window always shows the parameters of the active region — the current region is shown at the top (e.g. Region: Main Region (MASTER)).

DSP Tools window with analyzer, vectorscope and module rack
The DSP Tools window: at the top the FFT spectrum analyzer (20 Hz → 20 kHz) and the stereo vectorscope; below, the Stereo Link Control and the module rack, each module with an on/off switch, an arrow to expand it and a preset drop-down. Here the Noise Reduction module is expanded, showing its knobs.

How the window is laid out

  • Analyzer at the top — a real-time FFT spectrum (with a frequency/level tooltip on mouse hover) and a vectorscope showing the stereo image. The Preview DSP button turns listening with the effects applied on and off.
  • Stereo Link Control — the Enable Equalizer Link L/R and Enable Compressor Link L/R checkboxes: when enabled, changes to one channel are automatically applied to the other.
  • Module rack — the stack of effects, each with: an on/off switch (the small circle on the left), an arrow to expand/collapse the panel, and a preset drop-down. Note that Equalizer and Compressor are split into separate Left and Right panels.
FFT and audio mode The analyzer warns: “FFT analysis is active, but results are not reliable in Shared Mode. Use Exclusive Mode for accurate, bit-perfect analysis.” In WASAPI Shared mode the spectrum is only indicative; for an accurate reading switch to WASAPI Exclusive or ASIO in Audio Device.

Module order

The recommended order (“PRO order”), from top to bottom, is the order in which the signal is processed:

#ModuleIts role in a word
1Noise ReductionBackground hiss cleanup
2De-EsserSibilance control
3EqualizerTonal balance
4CompressorDynamics control
—AmbientSense of space/air
5SoftClipperSaturation and soft limiting
6GainFinal volume

At the top of the list you will also find the “fixed” modules Stereo Widener and Stereo Linking (L/R channel linking), which act on the overall stereo image.

How to use a module

  • Turn the module on/off with its switch: you can compare “before and after” on the fly.
  • Preset — choose a starting point from the preset drop-down (each module has presets designed for typical cases).
  • Parameters — adjust the knobs/values; with loop playback on, you hear the change immediately.
  • Expand/collapse the module panel to make room; the open/closed state is remembered.
Technical deep dive Most modules expose separate Left and Right channels, so you can treat them independently. When linking is enabled (e.g. CompLink for the compressor, or the general Stereo Linking) a change on one channel is mirrored on the other, to keep the stereo image consistent. The startup presets chosen for each module and the order of the chain are saved in the user settings and reloaded at the next launch.

In the pages that follow, each module has a card describing what it is for, how to set it up, a table of its parameters with their ranges, and the factory presets.

Noise Reduction Module 1

Reduces constant background noise — hiss, hum, microphone noise — by tracking the noise level and lowering the signal when it drops below that threshold. It is the first link in the chain because it is best to clean up before equalizing and compressing.

How to set it up

Start from a preset (Light / Medium / Strong Clean) and loop a part with only noise and a part with the voice. Raise the Sensitivity until the background disappears, but stop as soon as the voice starts to sound “muffled” or to open and close unnaturally. The Reduction sets how deep the cut goes.

ParameterRangeEffect
Sensitivity0 – 100How aggressively the noise is tracked: higher = kicks in sooner.
Reduction0 – 30 dBMaximum attenuation applied below the threshold.
AttackmsHow quickly the reduction kicks in when the signal drops.
ReleasemsHow quickly the reduction releases when the signal returns.

Factory presets

PresetSensitivityReductionAttackRelease
Light Clean358 dB15 ms180 ms
Medium Clean5012 dB20 ms220 ms
Strong Clean6518 dB25 ms260 ms
Technical deep dive The module continuously estimates the noise floor with a slow tracker (~150 ms) and compares the instantaneous envelope of the signal, in dB, with that threshold. Attack and Release shape the gain curve to avoid “pumping”. It is not spectral subtraction like FFT denoisers: it works on amplitude, so it is fast and transparent on constant hiss, less suited to complex tonal noise.

De-Esser Module 2

Softens sibilance — the “s”, “sh” and “z” sounds that become harsh and piercing, especially on close-miked voice. It acts only on the offending frequency band and only when that band exceeds a threshold.

How to set it up

  1. Enable Solo Sibilants to hear only the band the de-esser is “watching”: move the Frequency until you clearly hear the “s” sounds.
  2. Adjust the Bandwidth to narrow or widen that band around the sibilants.
  3. Turn Solo off, then lower the Threshold until the “s” sounds soften, and use the Ratio to decide how much to reduce them.
ParameterRangeEffect
Threshold−60 … 0 dBThreshold above which the sibilant is attenuated.
Ratio1 … 10Amount of reduction (higher = more strongly damped sibilant).
Frequency3,000 … 12,000 HzCenter of the sibilance band.
Bandwidth500 … 6,000 HzWidth of the controlled band.
Solo Sibilantson/offListen to the band in isolation, to center it.

Factory presets

PresetThresholdRatioFrequencyBandwidth
SoftVoice−25 dB2.55,500 Hz2,500 Hz
FemaleVocal−28 dB3.56,500 Hz3,000 Hz
MaleVocal−24 dB3.05,000 Hz2,500 Hz
PodcastClean−22 dB2.06,000 Hz2,000 Hz
StrongSibilance−30 dB5.07,000 Hz3,500 Hz
Technical deep dive The de-esser uses a band-pass filter (BiQuad) as a “detector”: the Q factor of the band is derived from Frequency / Bandwidth and limited between 0.3 and 12. When the energy in the band exceeds the threshold, a reduction proportional to the ratio is applied, with gain smoothing so as not to create artifacts. It is a band-detection de-esser, typical of vocal plugins.

Equalizer Module 3

Corrects the tonal balance: removes boominess from the lows, adds presence and “air” to the highs, carves out annoying frequencies. It is a 5-band parametric equalizer, with each band configurable independently.

How to set it up

For each band, choose the filter type, the frequency, the gain (cut or boost) and the Q (width of the adjustment). The frequency is adjusted on a logarithmic scale, as in professional plugins, so you get fine resolution in the lows.

ParameterDescription
TypeFilter type: Peaking (bell), HighPass (cuts the lows below the frequency), LowPass (cuts the highs above the frequency) and others.
FrequencyCenter/cutoff frequency of the band (logarithmic scale).
Gain (dB)How much to boost (+) or attenuate (−) around that frequency.
QWidth of the adjustment: high Q = narrow, surgical adjustment; low Q = broad, musical curve.

Factory presets

The included presets are: LowEnd Cut (cleans up excess lows), Bass Boost, Treble Boost, Air Boost (opens up the highs), Voice Clean, Podcast Voice (low cut + presence on speech), Telephone Effect (“telephone” voice), HiFi Smile Curve (boosted lows and highs) and Mastering Gentle Curve (a gentle mastering smile curve). Each preset sets the 5 bands to ready-made values that you can then fine-tune.

Tip On voice, a HighPass around 80 Hz removes boominess and handling noise without taking away body. Add a slight +2/+3 dB between 2.5 and 6 kHz for presence, and a touch of top end above 10 kHz for air. A few broad adjustments are better than lots of narrow bells.
Technical deep dive Each band is a BiQuad filter (second-order IIR). The frequency knob value is mapped logarithmically between a minimum and a maximum (freq = fMin·(fMax/fMin)^knob), while gain and Q are linear. The presets define frequency, gain, Q and type for each of the 5 bands — for example “Podcast Voice” combines a HighPass at 80 Hz, a cut at 150 Hz, a presence boost at 2.5 kHz and a safety LowPass.

Compressor Module 4

Evens out the dynamics: it reduces the gap between loud and soft parts, making the voice more consistent and present and the mix more “compact”. It is the module with the most parameters, with control over the knee, makeup and output ceiling.

How to set it up

  1. Lower the Threshold until you hear the compressor working on the peaks.
  2. Set a moderate Ratio (2:1–4:1 for voice) and adjust Attack/Release: a slower attack lets transients through, a suitable release avoids “pumping”.
  3. Recover the lost volume with Makeup and protect the output with the Ceiling.
ParameterRangeEffect
Threshold−60 … 0 dBLevel above which compression starts.
Ratio1 … 30Compression ratio (4 = 4:1). Higher = firmer.
Knee0 … 1“Knee”: 0 = abrupt action (hard knee), 1 = gradual onset (soft knee).
Attack0.1 … 200 msHow quickly it reacts to peaks.
Release5 … 2000 msHow quickly it releases after the peak.
Makeup−12 … +24 dBMakeup gain after compression.
Ceiling−12 … 0 dBMaximum output ceiling.

Factory presets

PresetThresholdRatioMakeupTypical use
SoftVocal−24 dB2.2+3.5 dBDelicate voice, light compression
Podcast−20 dB3.2+3.5 dBSteady, present speech
MusicPunch−18 dB3.8+2.5 dBMusic with more “punch”
HardLimiter−6 dB200 dBFirm peak limiting
MasteringGlue−12 dB2.0+2 dBGentle mastering “glue”
Technical deep dive Detection is dual-stage RMS (~50 ms fast + ~120 ms slow) for a natural envelope, with ~2 ms lookahead to anticipate transients and a DC high-pass filter at ~10 Hz. The knee is expressed as ±(knee·6 dB) around the threshold; gain smoothing uses two poles (~8 and ~25 ms). The safety saturation at the ceiling uses a pre-computed tanh (LUT). With CompLink linking enabled, Left and Right share the same parameters.

Ambient Effect

Adds a sense of space and air around the sound, like a small room/short reverb. Used sparingly it gives depth to the voice; pushed harder, it creates a “dreamy” effect.

ParameterEffect
MixAmount of effect blended into the dry sound (0 = none).
DelayDelay that defines the perceived “size” of the space (ms).

Factory presets

PresetMixDelay
Soft10%5 ms
Warm18%7.5 ms
Wide Room25%10 ms
Airy35%12.5 ms
Dreamy55%20 ms

SoftClipper Module 5

Adds warmth and “glue”, and at the same time acts as a soft limit on peaks: instead of clipping abruptly (hard clip), it rounds off the crests, gaining perceived loudness without unpleasant distortion.

ParameterEffect
DriveHow hard to “push” the signal into saturation (more drive = more warmth/density).
OutputOutput level, to compensate for the increase in volume.
CeilingMaximum ceiling of the output signal.

Factory presets

PresetDriveOutputCeiling
Gentle2 dB−1 dB−1 dB
Warm4 dB−2 dB−1 dB
Punch6 dB−3 dB−1 dB
Hard9 dB−4 dB−1 dB
Warning The soft clipper is an “additive” effect: at high drive settings the saturation becomes audible. On spoken voice use Gentle/Warm; keep Punch/Hard for musical material where you want density and aggression.

Gain Module 6

The last link: it sets the final volume of the region and can apply a fade-in and fade-out at module level. This is where you set the final level before export or LUFS normalization.

ParameterEffect
Gain (dB)Overall gain or attenuation.
Fade In / Fade OutFade-in/fade-out expressed in samples.

Factory presets

PresetGainFade InFade Out
Unity0 dB00
Boost Smooth+6 dB20002000
Cut Smooth−6 dB20002000
Boost Hard+12 dB500500
Cut Hard−12 dB500500

Stereo Widener & Stereo Linking Stereo image

The Stereo Widener widens the stereo image, giving the sound a sense of spaciousness and “width”. Stereo Linking is the switch that links the Left and Right channels of the modules, so a change on one side applies to both.

Parameter (Widener)Effect
DelayInter-channel delay that creates the widening (ms).
MixAmount of effect applied.

Factory presets

PresetDelayMix
Soft5 ms100%
Medium10 ms100%
Hard40 ms100%
Watch out for mono compatibility Delay-based stereo widening can weaken the signal when the audio is played back in mono (some sound systems, Bluetooth speakers, phones). If your material also has to hold up in mono, check by listening and use moderate presets.

⑪LUFS & loudness

Measuring and normalizing perceived loudness to the EBU R128 standard.

LUFS (Loudness Units Full Scale) measures how loud audio is to the human ear, not just technically. It is the yardstick streaming platforms use to match volumes: if your track is too loud, Spotify or YouTube turn it down; if it is too quiet, it falls behind the others. The LUFS Normalization window analyzes the file and tells you exactly where you are and how much gain you need to hit a target.

LUFS Normalization window with graph and batch module
The LUFS Normalization window: on the left the data (Integrated, Target, Gain to Target/Allowed/APPLY, RMS, Peak, PLR, Ceiling), on the right the LUFS Graph with the target and integrated lines. Below: Analyze/Preview/Apply/Undo, the preset/target/ceiling selection and the multi-file batch module.

The window and its commands

Open it from the Normalize LUFS button or from Tools ▸ Loudness Normalization (LUFS)….

CommandWhat it does
Analyze LUFSMeasures the track and fills in the data and the graph.
Preview LUFSPlays a preview with the normalization gain applied.
Apply NormalizationApplies the gain to the track.
Undo NormalizationUndoes the applied normalization.
Preset / Target LUFS / Ceiling (dBTP)Choose the target preset or set the target and ceiling by hand.

Target presets (LUFS)

PresetTargetCeilingTrue Peak Limiter
StreamingStandard−14 LUFS−1.0 dBON
YouTube−14 LUFS−1.5 dBON
Spotify−14 LUFS−2.0 dBON
AppleMusic−16 LUFS−1.0 dBON
BroadcastEBU−23 LUFS−1.0 dBON
LoudnessWar−9 LUFS0 dBOFF
NoLimiter−14 LUFS0 dBOFF
TruePeakOnly0 LUFS−1.0 dBON

Analyzing and normalizing

  1. Press Analyze LUFS: the program measures the track (ITU-R BS.1770 mono downmix) and draws the graph.
  2. Choose a target preset, or set Target LUFS and Ceiling by hand.
  3. Read Gain to Target (how much would be needed) and Gain Allowed (how much is permitted without exceeding the ceiling): Gain APPLY is the smaller of the two. Listen with Preview LUFS, then Apply Normalization.
MeasurementMeaningReference values
Integrated LUFSAverage loudness of the whole track (the main EBU R128 measurement).Streaming ≈ −14 · Podcast ≈ −16 · Broadcast TV −23 · Cinema −27
Short-Term LUFSLoudness over 3 s windows (trend section by section). Min = quietest part, Max = loudest.—
Momentary LUFSLoudness over 400 ms windows (very responsive).—
LRA (Loudness Range)Perceived dynamics = Short-Term Max − Min.Compressed pop 2–4 · Rock 4–8 · Jazz 8–14 · Classical 14+
RMSAverage energy (technical reference).Music −9/−12 dBFS · Speech −15/−20 dBFS
Peak (dBFS)Maximum digital sample peak.Correct master ≤ −1.0 dBFS
True Peak (dBTP)Actual peak after analog reconstruction (oversampling): the real risk of clipping on DACs and streaming.Streaming ≤ −1.0 dBTP
PLR (Peak-Loudness Ratio)Peak − LUFS: an index of dynamic “breathing room”. Low = heavily compressed.< 8 critical · 8–12 caution · > 12 good
Gain to TargetRequired gain = Target − Integrated.e.g. −18 → target −14 = +4 dB

Batch normalization

At the bottom of the window, the NORMALIZE MULTIPLE FILES (LUFS) module normalizes several files at once with the same preset: add the files with Add Files… (or empty the list with Clear List), choose the output folder with Browse, then Start Batch (or Stop Batch). A Log and progress bars track the process. Handy for evening out the volume of an entire collection.

Technical deep dive Integrated LUFS is calculated on the ITU-R BS.1770 mono downmix (channels summed with a 0.707 coefficient), not as an average of the individual channels or regions — this is the correct way a mastering engineer measures loudness. The normalization gain applied is always min(GainToTarget, GainAllowed), so raising the volume never exceeds the ceiling you set. When the preset has the True Peak Limiter enabled, a limiter with 4× oversampling (typical threshold −1.0 dBTP, with knee and lookahead) holds back the true peaks that would exceed the ceiling, avoiding distortion on DACs and streaming platforms.
In practice For a podcast aim for −16 LUFS with true peak ≤ −1 dBTP. For streaming music, −14 LUFS is the reference standard: going much louder will not make you sound louder on the platform, it will only cost you dynamics.

⑫Sound Analyzer

Measuring the quality of your track and comparing it with a reference track.

Open the Sound Analyzer from the button of the same name or from Processing ▸ Sound Analyzer…. It is not (just) a spectrum: it is a measurement station that, when you press Process And Analyze, calculates a set of indicators on the selected region and lets you compare them with a reference track that sounds “the way you want”.

Sound Analyzer window with the dynamics meters and the per-band analysis
The Sound Analyzer: at the top the dynamics meters (PLR, RMS, CF, TR, DR, LUFS, MMC) with their legend; below, the analysis of the loaded file by frequency band compared with the reference. At the bottom, the Save/Load Reference, COMPARE and Generate Report commands.

The dynamics meters

AbbreviationNameWhat it indicates
PLRPeak-Loudness RatioHow compressed/dynamic the track is: high PLR = more “breathing room”, low = more compressed.
RMSAverage energyAverage loudness and density: high RMS = denser mix, low = more open and clean.
CFCrest FactorStrength of transients relative to the average level: high = lively transients, low = flattened.
TRTransient RatioPresence and clarity of attacks: high = strong impact, low = soft/squashed attacks.
DRDynamic RangeHow much the track “breathes” (overall dynamics).
LUFSIntegrated LUFSAverage perceived loudness (see the dedicated chapter).
MMCLow-Level Energy (LLE)Minimum level of the musical content: the useful “floor” of the signal.

Per-band analysis and comparison

Below the meters, the Loaded File Analysis panel breaks the track down into bands — Sub, Bass, LowMid, Mid, HighMid, High — and reports Fast LUFS, PLR, Low-Level Energy and Peak Sample. The Reference panel next to it shows the same values for the reference track, so you can see at a glance where your mix has too much or too little.

CommandWhat it does
Process And AnalyzeRuns the analysis on the current region and updates the meters and bands.
Save As ReferenceSaves the current analysis as a .ccaref reference file.
Load ReferenceLoads a saved reference into the Reference panel.
COMPARECompares the loaded track and the reference, showing the deviations (Δ) band by band.
Generate ReportProduces the complete Mastering Report of the comparison.
How to use it in practice Take a commercial track whose sound you like, load it and press Save As Reference. Then go back to your track, press Process And Analyze and COMPARE: the deviations tell you, band by band, what to adjust in the EQ and compressor to get closer to that sound.

The FFT spectrum

The actual spectrum analyzer (20 Hz → 20 kHz, with a frequency/level tooltip) is built into the top of the DSP window, together with the stereo vectorscope. Its typical parameters:

ParameterWhat it controlsNotes
FFT SizeSamples used for the transform (512 → 8192).Larger = more frequency resolution but slower updates.
WindowWindow: Hann, Hamming, Blackman, Rectangular.Hann is the best compromise; Rectangular = more leakage.
Frequency ResolutionSpacing between the “bins”.= SampleRate / FFTSize (e.g. 44100/4096 ≈ 10.7 Hz).
Magnitude (dBFS)Amplitude in decibels; 0 dBFS = digital maximum.Typical: −20 / −120 dBFS.
Peak HoldHolds the maximum of each bin.Off · Short · Long · Infinite.
AveragingTime averaging of the spectrum.None (responsive) · Fast · Slow (stable).
Frequency scaleLogarithmic or linear.Log = like the ear, standard in pro plugins.
SmoothingSmooths the spectrum (1/3, 1/6, 1/12 octave).Useful for music analysis.
When in doubt The in-app help windows (FFT Analyzer – Help and LUFS Analyzer – Help) keep the parameter guide always at hand.

⑬Mastering Report

A text report that sums up the analysis of the track.

The Mastering Report gathers the technical information and the suggestions about the current track in a single screen, organized into sections:

SectionContents
Main InfoFile name, technical data, region info and DSP status.
NotesReadable mastering notes (comparison with the references).
Engineer ModeDetailed “sound engineer” technical notes.
AI ChainSuggested processing chain.
Text ReportComplete text report with the comparisons.
Dynamics CurveDynamics trend + dedicated notes.

The report's Export button copies the entire text to the clipboard, ready to paste into a document or an email.

⑭Exporting

Saving the result with the DSP chain applied.

Exporting produces the final file with all the processing applied (each region's DSP, fades, any normalization). Before you proceed, the summary window (Export Summary) shows what will be exported: for each region its range and active DSP chain, or — if there are no cuts — the whole main region.

SettingOptions
FormatMP3 or WAV
MP3 modeCBR (constant bitrate) or VBR (variable)
MP3 bitratefrom 96 to 320 kbps
Stereo WidenerYou can include or exclude it in the exported file
  1. File ▸ 💾 Save As… (or the export flow) opens the summary.
  2. Check the format, mode and bitrate; these values have defaults in the settings.
  3. If needed, open the ID3 editor from the summary to fill in the tags before saving.
  4. Confirm the export: a progress bar tracks the process.
Technical deep dive When there are several regions, each one is rendered with its own DspConfig and concatenated; with no cuts, the range of the main region (the start/end markers) is exported. The output formats are MP3 (LAME encoder, CBR at the chosen bitrate or a VBR preset) and WAV (16-bit PCM): the save dialog offers one or the other based on the default format. The format, the CBR/VBR mode and the bitrate are read from the preferences (DefaultExportFormat, DefaultMp3Mode, DefaultMp3Bitrate) and can be changed in the settings. ID3 tags are written to MP3 files only.

⑮ID3 tag editor

The MP3 metadata: title, artist, album, cover art.

From File ▸ ✏️ Edit mp3 Info… (or from the export summary) you open the ID3 tag editor. Here you fill in the information that music players display: title, artist, album, year, genre and — depending on the build — the cover art. Filling in the tags is especially useful for podcasts and collections, where files are organized and displayed based on this data.

⑯Projects

Saving your work to pick it up later.

A project stores not only the audio but also your editing work: markers, regions, fades and the DSP chains of each region. That way you can close and pick up exactly where you left off.

Command (Project menu)What it does
📦 Open Project…Opens a saved project.
📥 Save Project…Saves the current working state.
💾 Save Project As Single File…Saves everything (audio + settings) in a single self-contained file, handy for archiving or moving.
Tip Use Save Project As Single File when you need to move your work to another computer or back it up: you have just one file to manage, with no risk of losing the linked audio.

⑰System recorder

Capturing the audio coming out of your computer (Tools ▸ Recorder…).

The recorder captures system audio — what you hear from your speakers: a live stream, a video, an online radio station — through the sound card's loopback. Press Record to start and Stop to finish; a level meter shows the incoming signal.

WASAPI exclusive mode If you chose WASAPI Exclusive in the audio settings, recording system audio may not work: in exclusive mode the device is locked and other apps (browser, YouTube…) cannot play sound. To record system audio, use Shared mode. The program warns you in this case.

⑱Signal generator

Test tones and noise (Tools ▸ Tone Generator…).

The generator creates test signals, useful for calibrating a sound system, testing filters or creating reference material. Choose the signal type, then set the frequency, amplitude (in dB) and duration; for some types you can choose the stereo mode (Decorrelated, Phase Offset, Haas Delay).

Available types
Sine · White Noise · Pink Noise · Sweep · IMD Test · IMD 19+20 kHz · Tone Burst Stereo · Random Burst Stereo · Square Wave · Saw Wave · Triangle Wave
Technical deep dive The IMD (intermodulation distortion) tests and the classic 19+20 kHz test are used to measure the linearity of the audio chain; the Sweep lets you derive the frequency response; Pink Noise is the typical reference for equalizing a listening room because it has equal energy per octave.

⑲Voice & Text

Speech synthesis and transcription (Voice&Text menu).

Text-to-Speech (from text to voice)

With Voice&Text ▸ Text-to-Speech Module ▸ System Voice… you generate spoken audio from text, using the synthesis voices installed in Windows. If the computer has no usable voices, the program prompts you to install one from the Windows language settings.

Technical deep dive Both SAPI5 voices and the modern OneCore voices of Windows 10/11 are used. To add new voices/languages: Windows Settings ▸ Time & Language ▸ Speech.

Speech-to-Text (from voice to text)

With Voice&Text ▸ Speech-to-Text Module ▸ Convert Speech to Text you transcribe the portion between the start and end markers into text. The result appears in the Transcript window, with word timings when available.

Technical deep dive Transcription uses the offline Vosk engine with the model in the Models/en folder (English). The audio is converted to mono and resampled to 16 kHz before recognition. If the model is missing, the program reports that it was not found: it must be installed in the folder indicated. Since it works offline, transcription needs no connection, but it depends on the quality of the model and of the recording.

⑳Audio settings

Choosing the driver and device (Audio Device menu) and the default values (Settings menu).

From the Audio Device menu you select the output driver and device. From the Settings menu, instead, you set the default values: Export Settings… (MP3/WAV format, CBR/VBR mode, bitrate) and DSP Settings… (startup presets and chain order). The choice of driver affects latency and compatibility:

DriverCharacteristics
WASAPI — SharedShared mode: coexists with other apps. It is the right choice if you also want to record system audio.
WASAPI — ExclusiveExclusive access to the device: lower latency and full quality, but it blocks audio from other apps and prevents system recording.
ASIOVery low-latency driver (professional sound cards). Always exclusive; requires an installed ASIO driver.
Which one to choose For normal editing and listening, WASAPI Shared is the safest and most compatible choice. Switch to ASIO only if you have a professional sound card and need the lowest latency.

⌨Keyboard shortcuts

The complete reference — keep it handy.

Playback and markers

KeyAction
SpacePlay / Stop
+Insert a cut marker (creates a region)
−Remove the cut marker
Ctrl+Shift+XRemove all cut markers

Zoom and scrolling

Key / gestureAction
Mouse wheelZoom in / out
Shift + wheelScroll (pan)
Shift + dragHorizontal “accordion” zoom
↑ / ↓Zoom in / out
← / →Scroll left / right
OFull view (zoom all the way out)
PCenter the view on the playhead
SCenter the view on the start marker
ECenter the view on the end marker
IMove the playhead to the start marker

Editing

KeyAction
Ctrl+WInsert Silence (silences the selection)
Ctrl+RReverse (reverses the selection)
Ctrl+PPaste

Windows

KeyAction
F9Show / hide the DSP window
F10Show / hide the TIMELINE overlay (markers, regions and cuts)
Reminder The navigation and marker shortcuts are always shown on the welcome screen when no file is loaded.

📖Glossary

The technical terms you will come across in the program, in brief.

TermMeaning
LUFSUnit of measurement of perceived loudness (EBU R128 standard). The yardstick streaming services use to match volumes.
dBFSDecibels relative to digital full scale; 0 dBFS is the maximum before clipping.
True Peak (dBTP)The true peak of the signal after conversion to analog: it can exceed the digital peaks and cause distortion.
HeadroomMargin between the signal level and 0 dBFS.
ThresholdLevel above which a processor (compressor, de-esser) starts to act.
RatioReduction ratio applied above the threshold (e.g. 4:1).
Attack / ReleaseHow quickly an effect kicks in and releases.
KneeHow gradual (soft) or abrupt (hard) the onset of compression is around the threshold.
QWidth of an EQ filter's action: high = narrow, low = broad.
SibilantsThe “s/sh/z” sounds, high-frequency energy that the de-esser softens.
PlayheadThe cursor that shows the playback position.
RegionPortion of audio between two cut markers, with its own DSP chain.
DSPDigital Signal Processing: the digital processing of sound (the effects in the chain).
CBR / VBRConstant / variable bitrate in MP3.
WASAPI / ASIOWindows audio driver systems; ASIO is low-latency, for pro sound cards.

🛟Troubleshooting

Quick answers to the most common situations.

I can't hear anything during playback

In the Audio Device menu, check that the output device is the right one. If you chose WASAPI Exclusive or ASIO and another app is holding the device, switch to WASAPI Shared.

The system recorder doesn't capture anything

It is almost always WASAPI Exclusive mode: it prevents other apps from playing sound, so there is nothing to record. Set WASAPI Shared and try again.

Transcription says “model not found”

The Vosk model is missing: it must be in the Models/en folder next to the program. Download the language model and place it there.

Text-to-Speech has no voices

Install a voice from the Windows settings (Time & Language ▸ Speech). The program sees both SAPI5 and OneCore voices.

After normalization the volume did not go up as much as expected

The gain applied is the smaller of Gain to Target and Gain Allowed: if raising the level all the way to the target would exceed the ceiling, the program stops short so as not to distort. To go higher, you first need to reduce the peaks (compressor / soft clipper).

The voice sounds “muffled” after cleanup

Probably the Noise Reduction is too aggressive: lower Sensitivity and/or Reduction. Noise reduction should only be pushed until the background disappears, no further.

The file sounds good on headphones but weak in mono

Check the Stereo Widener: delay-based widening can weaken the signal in mono. Reduce the mix or turn it off for export if the material has to hold up in mono.


CleanCutAudio user manual — reference document generated from the application's features. The numeric values (parameter ranges and factory presets) reflect the program's current configuration and may change in later versions.